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webrtc/modules/audio_processing
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Per Åhgren ef5d5af3a0 AEC3: Increasing the accuracy of the detection for early reverb 
This CL introduces an adaptive estimation of the early reverb
in the estimation for the room reverberation. The benefits of
this is that for room with long early reflections there is
a lower risk of underestimating the reverberation.

This CL is for a landing the code in
https://webrtc-review.googlesource.com/c/src/+/87420,
and the review of the code was done in that CL. The author of
code is devicentepena@webrtc.org

Bug: webrtc:9479, chromium:865397
Change-Id: Id6f57e2a684664aef96e8c502e66775f37da59da
Reviewed-on: https://webrtc-review.googlesource.com/91162
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24146}
2018-07-30 22:34:19 +00:00
..
aec Delete root header file typedef.h. 2018-07-25 14:59:26 +00:00
aec3 AEC3: Increasing the accuracy of the detection for early reverb 2018-07-30 22:34:19 +00:00
aec_dump Use absl::make_unique and absl::WrapUnique directly 2018-07-05 10:59:49 +00:00
aecm Delete root header file typedef.h. 2018-07-25 14:59:26 +00:00
agc Move fft4g to proper third_party directory 2018-07-25 15:44:53 +00:00
agc2 Delete root header file typedef.h. 2018-07-25 14:59:26 +00:00
audio_generator Add stub draft of audio generator to APM 2018-03-05 09:28:52 +00:00
echo_detector Replace rtc::Optional with absl::optional in modules/audio processing 2018-06-19 10:38:56 +00:00
include Remove old temporary webrtc::PostProcessing typedef 2018-07-27 15:43:57 +00:00
intelligibility Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
logging Remove stringstream usages from the APM 2018-04-06 14:17:03 +00:00
ns Move fft4g to proper third_party directory 2018-07-25 15:44:53 +00:00
test Delete root header file typedef.h. 2018-07-25 14:59:26 +00:00
transient Move fft4g to proper third_party directory 2018-07-25 15:44:53 +00:00
utility Delete root header file typedef.h. 2018-07-25 14:59:26 +00:00
vad Move fft4g to proper third_party directory 2018-07-25 15:44:53 +00:00
audio_buffer.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
audio_buffer.h Delete root header file typedef.h. 2018-07-25 14:59:26 +00:00
audio_buffer_unittest.cc
audio_frame_view_unittest.cc Add namespace 'webrtc' to AudioFrameView. 2018-05-14 12:33:49 +00:00
audio_processing_impl.cc Delete root header file typedef.h. 2018-07-25 14:59:26 +00:00
audio_processing_impl.h Revert "Add one-stop-shop for built-in AEC toggling in APM" 2018-07-23 14:48:17 +00:00
audio_processing_impl_locking_unittest.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
audio_processing_impl_unittest.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
audio_processing_performance_unittest.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
audio_processing_unittest.cc Delete root header file typedef.h. 2018-07-25 14:59:26 +00:00
BUILD.gn Move fft4g to proper third_party directory 2018-07-25 15:44:53 +00:00
common.h
config_unittest.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
debug.proto Options and settings for the Pre-amplifier. 2018-04-16 12:25:48 +00:00
DEPS
echo_cancellation_bit_exact_unittest.cc
echo_cancellation_impl.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
echo_cancellation_impl.h Enforcing a stream delay of 0 to be assumed in the AEC on Chrome OS 2017-12-22 15:42:13 +00:00
echo_cancellation_impl_unittest.cc Use AudioProcessingBuilder everywhere AudioProcessing is created. 2018-01-09 13:45:20 +00:00
echo_control_mobile_impl.cc Turn off comfort noise generation by default in AECM 2018-07-24 08:52:36 +00:00
echo_control_mobile_impl.h
echo_control_mobile_unittest.cc
gain_control_for_experimental_agc.cc
gain_control_for_experimental_agc.h
gain_control_impl.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
gain_control_impl.h Replace rtc::Optional with absl::optional in modules/audio processing 2018-06-19 10:38:56 +00:00
gain_control_unittest.cc
gain_controller2.cc Make possible to activate adaptive AGC2 in the APM. 2018-03-29 09:42:07 +00:00
gain_controller2.h Make possible to activate adaptive AGC2 in the APM. 2018-03-29 09:42:07 +00:00
gain_controller2_unittest.cc Set a positive initial gain in the Adaptive Digital GC. 2018-04-27 09:05:25 +00:00
level_estimator_impl.cc
level_estimator_impl.h
level_estimator_unittest.cc
low_cut_filter.cc
low_cut_filter.h
low_cut_filter_unittest.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
noise_suppression_impl.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
noise_suppression_impl.h
noise_suppression_unittest.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
OWNERS Adding alessiob@ and minyue@ as owners of APM. 2018-07-02 07:45:31 +00:00
render_queue_item_verifier.h
residual_echo_detector.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
residual_echo_detector.h Add more parameters to the Initialize function of the echo detector. 2018-03-15 09:21:56 +00:00
residual_echo_detector_unittest.cc Change echo detector to scoped_refptr 2018-06-14 09:51:41 +00:00
rms_level.cc Replace rtc::Optional with absl::optional in modules/audio processing 2018-06-19 10:38:56 +00:00
rms_level.h Delete root header file typedef.h. 2018-07-25 14:59:26 +00:00
rms_level_unittest.cc Move some more numeric utility code from rtc_base/ to rtc_base/numerics/ 2017-11-22 12:39:39 +00:00
splitting_filter.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
splitting_filter.h
splitting_filter_unittest.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
three_band_filter_bank.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
three_band_filter_bank.h
typing_detection.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
typing_detection.h Delete root header file typedef.h. 2018-07-25 14:59:26 +00:00
voice_detection_impl.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
voice_detection_impl.h
voice_detection_unittest.cc